[asterisk-commits] kmoore: testsuite/asterisk/trunk r3494 - in /asterisk/trunk/tests/channels/SI...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Oct 19 13:45:12 CDT 2012
Author: kmoore
Date: Fri Oct 19 13:45:03 2012
New Revision: 3494
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=3494
Log:
Fix IPv6 SIP attended transfer test
This test had been bouncing a while back and so was disabled until we
got a chance to look into the matter. This test has essentially been
rewritten using SIPp's extended 3rd party call control protocol and the
configuration-driven framework versus the previous implementation using
pjsua and run-test python script.
(closes issue SWP-4661)
Review: https://reviewboard.asterisk.org/r/2147/
Reported-by: Matt Jordan
Patch-by: Kinsey Moore
Added:
asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/hangup_call.py (with props)
asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/sipp/
asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/sipp/slave_cfg.conf (with props)
asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/sipp/uac-call.xml (with props)
asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/sipp/uac-calls-and-refer.xml (with props)
asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/sipp/uas-no-hangup.xml (with props)
Removed:
asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/run-test
Modified:
asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/configs/ast1/sip.conf
asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/test-config.yaml
Modified: asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/configs/ast1/sip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/configs/ast1/sip.conf?view=diff&rev=3494&r1=3493&r2=3494
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/configs/ast1/sip.conf (original)
+++ asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/configs/ast1/sip.conf Fri Oct 19 13:45:03 2012
@@ -1,6 +1,7 @@
[general]
directmedia=no
bindaddr=[::1]:5060
+pedantic=no
[end_a]
context=transfertest
@@ -28,3 +29,13 @@
insecure=invite
disallow=all
allow=ulaw
+
+[end_d]
+context=transfertest
+type=friend
+host=::1
+port=5068
+insecure=invite
+disallow=all
+allow=ulaw
+
Added: asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/hangup_call.py
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/hangup_call.py?view=auto&rev=3494
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/hangup_call.py (added)
+++ asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/hangup_call.py Fri Oct 19 13:45:03 2012
@@ -1,0 +1,16 @@
+#!/usr/bin/env python
+'''
+Copyright (C) 2012, Digium, Inc.
+Kinsey Moore <kmoore at digium.com>
+
+This program is free software, distributed under the terms of
+the GNU General Public License Version 2.
+'''
+
+
+def hangup_call(ami, event):
+ '''This hangs up the last remaining call path for the IPv6 attended
+ transfer test, causing the test to end instead of waiting for it to time
+ out after 30 seconds.'''
+ ami.hangup("SIP/end_b-00000001")
+ return True
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Added: asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/sipp/slave_cfg.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/sipp/slave_cfg.conf?view=auto&rev=3494
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/sipp/slave_cfg.conf (added)
+++ asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/sipp/slave_cfg.conf Fri Oct 19 13:45:03 2012
@@ -1,0 +1,2 @@
+s1;127.0.0.1:8081
+m;127.0.0.1:8082
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svn:keywords = Author Date Id Revision
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svn:mime-type = text/plain
Added: asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/sipp/uac-call.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/sipp/uac-call.xml?view=auto&rev=3494
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/sipp/uac-call.xml (added)
+++ asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/sipp/uac-call.xml Fri Oct 19 13:45:03 2012
@@ -1,0 +1,122 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="IPv6 Attended Transfer Slave">
+ <!-- wait for command from master to initiate call -->
+ <recvCmd src="startcall">
+ <action>
+ <ereg regexp="master-id: (.*)" search_in="msg" check_it="true" assign_to="1,remotecallid"/>
+ <strcmp assign_to="1" variable="1" value=""/>
+ </action>
+ </recvCmd>
+
+ <send>
+ <![CDATA[
+
+ INVITE sip:call_c@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/UDP [local_ip]:[local_port];rport;branch=z9hG4bKPjwCaFMEItTqvbbSHXiHoyHL.lpaZPFzeF
+ From: <sip:[local_ip]>;tag=WdXXeyLSHfy.tFproy3IQd2MNwJsezQW
+ To: sip:call_c@[remote_ip]
+ Contact: <sip:[remote_ip]:[local_port];ob>
+ Call-ID: [call_id]
+ CSeq: 18000 INVITE
+ Allow: INVITE, ACK, BYE, CANCEL, NOTIFY, REFER
+ User-Agent: SIPp
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 3556970359 3556970359 IN IP6 ::1
+ c=IN IP6 ::1
+ b=AS:84
+ t=0 0
+ a=X-nat:0
+ m=audio 40002 RTP/AVP 0
+ c=IN IP6 ::1
+ b=TIAS:64000
+ a=rtcp:40003 IN IP6 ::1
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true"
+ rtd="true"
+ crlf="true">
+ </recv>
+
+ <recv response="180"
+ optional="true"
+ rtd="true"
+ crlf="true">
+ </recv>
+
+ <recv response="183"
+ optional="true"
+ rtd="true"
+ crlf="true">
+ </recv>
+
+ <recv response="200"
+ rtd="true"
+ crlf="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:call_c@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/UDP [remote_ip]:[local_port];rport;branch=z9hG4bKPj9bTJ92liJq5dFYagMI6Fwb5t-580XGHQ
+ From: <sip:[local_ip]>;tag=WdXXeyLSHfy.tFproy3IQd2MNwJsezQW
+ To: sip:call_c@[remote_ip];tag=as1703baba
+ Call-ID: [call_id]
+ CSeq: 18000 ACK
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <sendCmd dest="m">
+ <![CDATA[
+ Call-ID: [$remotecallid]
+ From: callstarted
+ ]]>
+ </sendCmd>
+
+ <recv request="BYE"
+ rtd="true"
+ crlf="true">
+ </recv>
+
+ <sendCmd dest="m">
+ <![CDATA[
+ Call-ID: [$remotecallid]
+ From: startbye
+ ]]>
+ </sendCmd>
+
+ <recvCmd src="finishbye"/>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ Via: SIP/2.0/UDP [remote_ip]:[remote_port];rport=[remote_port];received=[remote_ip];branch=z9hG4bK6e3aa094
+ Call-ID: [call_id]
+ From: <sip:call_c@[local_ip]>;tag=as1703baba
+ To: <sip:[remote_ip]>;tag=WdXXeyLSHfy.tFproy3IQd2MNwJsezQW
+ CSeq: 102 BYE
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <timewait milliseconds="4000"/>
+
+ <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+ <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
+
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svn:keywords = Author Date Id Revision
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------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/sipp/uac-calls-and-refer.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/sipp/uac-calls-and-refer.xml?view=auto&rev=3494
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/sipp/uac-calls-and-refer.xml (added)
+++ asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/sipp/uac-calls-and-refer.xml Fri Oct 19 13:45:03 2012
@@ -1,0 +1,164 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="IPv6 Attended Transfer">
+ <send>
+ <![CDATA[
+
+ INVITE sip:call_b@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/UDP [local_ip]:[local_port];rport;branch=z9hG4bKPjwCaFMEItTqvbbSHXiHoyHL.lpaZPFzeF
+ Max-Forwards: 70
+ From: <sip:[local_ip]>;tag=WdXXeyLSHfy.tFproy3IQd2MNwJsezQW
+ To: sip:call_b@[remote_ip]
+ Contact: <sip:[remote_ip]:[local_port];ob>
+ Call-ID: [call_id]
+ CSeq: 17216 INVITE
+ Allow: INVITE, ACK, BYE, CANCEL, NOTIFY, REFER
+ User-Agent: SIPp
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 3556970359 3556970359 IN IP6 ::1
+ c=IN IP6 ::1
+ b=AS:84
+ t=0 0
+ a=X-nat:0
+ m=audio 40002 RTP/AVP 0
+ c=IN IP6 ::1
+ b=TIAS:64000
+ a=rtcp:40003 IN IP6 ::1
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true"
+ rtd="true"
+ crlf="true">
+ </recv>
+
+ <recv response="180"
+ optional="true"
+ rtd="true"
+ crlf="true">
+ </recv>
+
+ <recv response="183"
+ optional="true"
+ rtd="true"
+ crlf="true">
+ </recv>
+
+ <recv response="200"
+ rtd="true"
+ crlf="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:call_b@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/UDP [remote_ip]:[local_port];rport;branch=z9hG4bKPj9bTJ92liJq5dFYagMI6Fwb5t-580XGHQ
+ From: <sip:[local_ip]>;tag=WdXXeyLSHfy.tFproy3IQd2MNwJsezQW
+ To: sip:call_b@[remote_ip];tag=as1703baba
+ Call-ID: [call_id]
+ CSeq: 17216 ACK
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <sendCmd dest="s1">
+ <![CDATA[
+ Call-ID: callslave
+ From: startcall
+ master-id: [call_id]
+ ]]>
+ </sendCmd>
+
+ <recvCmd src="callstarted"/>
+
+ <send>
+ <![CDATA[
+
+ REFER sip:call_b@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/UDP [remote_ip]:[local_port];rport;branch=z9hG4bKPjY5k6ciCJOMSD7JrSQIrgqXFT-sNpZgc5
+ Max-Forwards: 70
+ From: <sip:[local_ip]>;tag=pqupJ2PptaapqbjgIbK5mg265SAi1pqn
+ To: sip:call_b@[remote_ip];tag=as5cb07cdc
+ Contact: <sip:[remote_ip]:[local_port];ob>
+ Call-ID: [call_id]
+ CSeq: 18217 REFER
+ Refer-To: <sip:call_b@[remote_ip]:[remote_port]?Replaces=callslave%3Bto-tag%3D%3Bfrom-tag%3DWdXXeyLSHfy.tFproy3IQd2MNwJsezQW>
+ Referred-By: <sip:[remote_ip]:[local_port]>
+ User-Agent: SIPp
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="202"
+ rtd="true"
+ crlf="true">
+ </recv>
+
+ <recv request="NOTIFY"
+ rtd="true"
+ crlf="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ Via: SIP/2.0/UDP [remote_ip]:[remote_port];rport=[remote_port];received=[remote_ip];branch=z9hG4bK6d34b376
+ Call-ID: [call_id]
+ From: <sip:call_b@[local_ip]>;tag=as5cb07cdc
+ To: <sip:[remote_ip]>;tag=pqupJ2PptaapqbjgIbK5mg265SAi1pqn
+ CSeq: 102 NOTIFY
+ Contact: <sip:[remote_ip]:[local_port];ob>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recvCmd src="startbye"/>
+
+ <send>
+ <![CDATA[
+
+ BYE sip:call_b@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/UDP [remote_ip]:[local_port];rport;branch=z9hG4bKPjTDMWblhklY8zxSCoWz4vAvkRwJVCVeey
+ From: <sip:[local_ip]>;tag=pqupJ2PptaapqbjgIbK5mg265SAi1pqn
+ To: sip:call_b@[remote_ip];tag=as5cb07cdc
+ Call-ID: [call_id]
+ CSeq: 18500 BYE
+ User-Agent: SIPp
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="200"
+ rtd="true"
+ crlf="true">
+ </recv>
+
+ <sendCmd dest="s1">
+ <![CDATA[
+ Call-ID: callslave
+ From: finishbye
+ master-id: [call_id]
+ ]]>
+ </sendCmd>
+
+ <timewait milliseconds="4000"/>
+
+ <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+ <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
+
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Added: asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/sipp/uas-no-hangup.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/sipp/uas-no-hangup.xml?view=auto&rev=3494
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/sipp/uas-no-hangup.xml (added)
+++ asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/sipp/uas-no-hangup.xml Fri Oct 19 13:45:03 2012
@@ -1,0 +1,78 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Basic UAS responder">
+ <recv request="INVITE" crlf="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 180 Ringing
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[pid]SIPpTag01[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[pid]SIPpTag01[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio [media_port] RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv request="ACK"
+ optional="true"
+ rtd="true"
+ crlf="true">
+ </recv>
+
+ <recv request="BYE">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <timewait milliseconds="4000"/>
+
+ <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+ <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
+
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svn:keywords = Author Date Id Revision
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svn:mime-type = text/plain
Modified: asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/test-config.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/test-config.yaml?view=diff&rev=3494&r1=3493&r2=3494
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/test-config.yaml (original)
+++ asterisk/trunk/tests/channels/SIP/sip_attended_transfer_v6/test-config.yaml Fri Oct 19 13:45:03 2012
@@ -1,17 +1,48 @@
testinfo:
- skip: 'Skip while failures are debugged'
- summary: 'Test SIP Attended Transfer'
+ summary: 'Test SIP Attended Transfer over IPv6'
description: |
- 'This test verifies the SIP_REFER with Replaces attended transfer routine.'
+ "This test verifies the SIP REFER with Replaces attended transfer routine over IPv6. Two calls are originated and terminated at pairs SIPp scenarios. After the calls are brought up, the two originating SIPp instances communicate using SIPp's 3PCC Extended Mode to pass the Call-ID of the originating leg of the second call to the originating leg of the primary call. The originating leg of the primary call uses this Call-ID in a REFER to bridge the two terminating legs. This bridged call is then hung up to terminate the test."
properties:
minversion: '1.8.0.0'
dependencies:
- python : 'twisted'
- python : 'starpy'
- - app : 'pjsua'
- custom : 'ipv6'
- - custom : 'pjsuav6'
+ - app : 'sipp'
tags:
- SIP
- - transfer
+ - transfer
+
+test-modules:
+ add-test-to-search-path: 'True'
+ test-object:
+ config-section: sipp-config
+ typename: 'sipp.SIPpTestCase'
+ modules:
+ -
+ config-section: ami-config
+ typename: 'ami.AMIEventModule'
+
+sipp-config:
+ test-iterations:
+ -
+ scenarios:
+ - { 'target': '[::1]', 'key-args': {'scenario': 'uas-no-hangup.xml', '-i': '[::1]', '-p': '5066'} }
+ - { 'target': '[::1]', 'key-args': {'scenario': 'uas-no-hangup.xml', '-i': '[::1]', '-p': '5067'} }
+ - { 'target': '[::1]', 'key-args': {'scenario': 'uac-call.xml', '-i': '[::1]', '-p': '5068', '-slave': 's1', '-slave_cfg': 'slave_cfg.conf'} }
+ - { 'target': '[::1]', 'key-args': {'scenario': 'uac-calls-and-refer.xml', '-i': '[::1]', '-p': '5065', '-master': 'm', '-slave_cfg': 'slave_cfg.conf'} }
+
+ami-config:
+ -
+ type: 'callback'
+ conditions:
+ match:
+ Event: 'Bridge'
+ Bridgestate: 'Unlink'
+ Bridgetype: 'core'
+ Channel1: '[sS][iI][pP]/end_a-.*'
+ Channel2: '[sS][iI][pP]/end_d-.*ZOMBIE>'
+ callbackModule: 'hangup_call'
+ callbackMethod: 'hangup_call'
+ count: '1'
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